Fiter design is the process of designing a filter.

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54 views

Reconstruct FIR frequency response from the coefficients

There is a lot of tools for designing FIR filters. Using one such tool I have designed a filter that satisfies my needs. It's coefficients are as follows: ...
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1answer
38 views

correct for shift from butterworth filter?

I have implemented a python butterworth filter but it seems to shift the data a lot, the x location of my data is very important to me for my analysis. Is there a way to correct for this shift ...
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1answer
136 views

Real and Imaginary Frequency Responses of a single complex pole

I'm filtering a real signal with a single complex pole with a complex coefficient (a_re [real part] and a_im [imaginary part]), I also have a gain coefficient but I'm gonna leave it out for the sake ...
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0answers
25 views

Using Python with scipy.signal firwin, how do I know how many taps/coefficients should be used?

I need to alter an FIR filter which I know very little about. I have a signal which need to be altered, and looking at a Python script used for this task, I see that it uses Scipy's signal library, ...
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1answer
38 views

Slepian or DPSS window

I am planning to use Slepian or DPSS window in my application where I want central lobe to be concentrated and also have low bandwidth: ...
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3answers
66 views

How to filter noise from continuous path in image

Problem statement In the dataset above, you can notice a continuous path and then a noise-band at around 320-400 on the x-axis. I want to be able to apply a filter that amplifies the continuous ...
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1answer
61 views

Low pass filter to maintain edge information

I am looking for a kernel acting as a low pass filter that satisfies these conditions: $$K(-\mathbf{u})=K(\mathbf{u}) \tag{1}$$ $$K(\mathbf{u}) \ge K(\mathbf{v}), \;\; \text{if} \;\; ...
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1answer
51 views

Design and implementation of causal band-pass filter for biosignals - what to consider? [closed]

I currently work in a project of clinical software (C#) which deals with clinical biosignals, specifically EMG and other low-frequency signals (load-cells, goniometers, and other biomechanical ...
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1answer
40 views

Approximate a system frequency response with a filter in Matlab

suppose I know the frequency response of a (linear) model approximating a real physical system but only at a specific frequency $f_0$ (so basically I have a complex number whose module is the ...
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0answers
57 views

Any way to compensate for fractional delay filter's weakness between taps?

A little background: I have an FPGA (running at 200MHz) that I am creating multipath events using buffers, all of that works. I take RF, mix down to IF, sample, make adjustments, out the DAC, and ...
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2answers
40 views

Low-pass vs median filter delay

I need to design a filter to reduce noise from accelerometer readings and use this data to a control system. One of the problems of a simple low-pass filter (like a single pole IIR filter) is the ...
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1answer
44 views

Designing the filtering for a time domain data acquisition application

I want to filter sensor signals for a data acquisition application on an ATmega microcontroller. The interesting parts of the signal are in the range 0-5Hz, and I'd like to leave these mostly ...
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1answer
79 views

Best way to inverse Hilbert on FPGA?

I have a real signal that is being sampled via ADC and passed to a Xilinx V6 FPGA. I then do a Hilbert transform on it using the FIR compiler core and get the complex signal (I and Q) to tweak the ...
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4answers
104 views

Why are recursives methods useful for FIR filter design?

As I understand it, FIR filtering is a linear process. That mean for me that the whole filtering process will have a fully predictable behavior. So, could someone explain why a universal ...
3
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1answer
63 views

Confusion about result of FIR All-pass filter design

I try to design a FIR All-pass filter with random phase in the frequency domain. I am a bit confused by my result and am not sure if the reason is a programming error or a misconception about ...
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1answer
39 views

Is there a way to break an analog sound signal up and send it through two radio chips at double the data rate of one?

Here is a radio chip for audio that sends at a CD-level data rate. Could you sample an analog signal (for instance from a guitar pick-up or microphone) in such a way that you could transmit using two ...
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1answer
50 views

Applying Discrete Filters

I have an exercise in DSP. I learnt about discrete filter but I don't know how to apply it to produce an output from an input signal. Please guide me to do this: Assume fundamental sample time is ...
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2answers
133 views

How do I manually plot the frequency response of a bandpass Butterworth filter in MATLAB?

I have code like below that applies a bandpass filter onto a signal. I am quite a noob at DSP and I want to understand what is going on behind the scenes before I proceed. To do this, I want to know ...
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1answer
56 views

FIR-Decimation and Low-pass filter (taps vs number of input points vs number of decimation stages)

I am new to digital filters and trying to understand parameters behind them. Scenario I am trying to get DC value from signals sampled at 100kHz sampling rate. such high frequency is used due to ...
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0answers
18 views

IIR Design with Neural Networks

i'm trying to apply this paper: using C++ Design of Digital Recursive Filter Using Artificial Neural Network In this paper the weights are the coefficients and they start with weight a(0) = 1 and ...
0
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1answer
75 views

First-order low pass filter with bounded derivative

Given a function $x(t)$ and a first order low-pass filter $\frac{dy}{dt} = \frac{1}{\alpha} (x(t) - y(t))$, is there a way to choose $\alpha$ analytically such that the derivative of $y(t)$ is ...
2
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1answer
77 views

How to remove historical values from exponential filter

Suppose I have two time series $A:=[0,0,0,4,5,6]$ and $B:=[1,2,3,4,5,6]$. I implement the following filter (with initial value at the first element): $$ f(t) = f(t-1) + \alpha*(f(t)-f(t-1))$$ with ...
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1answer
74 views

Low Pass Filter in AM Demodulation

I am trying to recover my baseband signal that was AM modulated. I am using the filter command in MATLAB to perform the low pass filtering ...the code is given below: ...
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2answers
89 views

Difference between Equiripple & Least Squares design for FIR digital filters

For an efficient and optimized digital FIR filter design, there are two methods available broadly, Equiripple filter design & Least Squares filter design. A general method for designing a filter ...
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2answers
72 views

Types of Filters

Which filters response is maximally flat at origin? It seems to me that both High Pass Filter and Band Pass Filter have maximum flat response at origin, although it is almost zero in case of Bandpass ...
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1answer
64 views

Interpret group delay of a filter

I keep reading that a constant group delay over a frequency band is desirable for a filter. But a constant group delay means a linear decrease in the phase delay, which would distort the signal. ...
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0answers
53 views

Signal leveling using scipy

Apologies if I'm using incorrect terms, and if this question's already been answered here. I have a data set which consists of a high DC signal modulated by AC signal (taken from custom-built ...
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2answers
53 views

Adaptive gaussian filter for noise image

I am looking for methods to enhance noisy images, where: some pixels in the image are very noise, some other pixels do not contain so much noise. My first thought is to build an adaptive Gaussian ...
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2answers
62 views

Why is beginning of this FIR filtered signal attenuated?

I apply an FIR filter passing the band around 50/1000 and attenuating the other three frequency components 30/1000, 70/1000 and 110/1000. Both filters are generated using the Remez algorithm. The ...
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2answers
102 views

Matlab - Creating a characteristic for Peak and Shelving filter

I have designed few shelving and peak filter. And I have calculated the a, b coeficients for several gains for these filters. Now I would like to create some kind of characteristic in Matlab so I ...
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2answers
31 views

Is it possible to calculate a z-transform for a filter calculated with Parks-McClellan?

I would like to know whether this can be done and if then how would this feat be acomplished in Matlab given a filter calculated with Remez? h = remez(...); If ...
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1answer
30 views

Trying to apply Parks-McClellan optimal FIR filter design

My goal is to apply the Remez exchange algorithm in R to design an FIR band pass filter for a specific frequency window. The question is not supposed to be referring to R of course but about how to ...
0
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1answer
89 views

Rate of fall of sidelobe level

Anyone can explain me about how measure side lobe roll-off rate of kaiser window from Spectrum in matlab where side lobe roll-off rate is the asymptotic decay rate, in decibels per octave of ...
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36 views

how does a karaoke work? [duplicate]

in audio signals the extraction of the background score , isolating the speech is known as the karaoke. if i use a simple band reject filter which will eliminate the spectral components in (0-4)KHz, ...
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0answers
40 views

Building a filter to allow Frequencies between 400 Hz to 4 KHz

I want to build a filter that allows 400Hz to 4KHz and strongly attenuates all other signals especially those from Cellphones (900 MHz to 2.5 GHz), DSL frequencies & Television VL, VHF & UHF ...
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0answers
56 views

Doppler shift using PLL

I am trying to apply phase lock loop to calculate phase shift due to doppler effect and finally calculate the doppler frequency($\Delta f$). I am assuming that the received wave is of the form, $r(t) ...
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4answers
215 views

Digital Anti Aliasing Filter For Waveform Band Limiting

I am working on some waveform generation software and I am trying to figure out what the best type of filter would be to use a an Anti Aliasing filter for my algorithm. I am generating my waves in a ...
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1answer
50 views

Why multiplying frequency kernel with sinusoid results shifting frequency domain

I am having great time while reading "The Scientist and Engineer's Guide to Digital Signal Processing" book. In "Chapter-14 Introduction to Filters" author indicates we can create high pass filter ...
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1answer
93 views

Low Pass FIR Filter [closed]

I want to design FIR low-pass filter with the following specifications: Cut-off frequency: 200 Hz Order: 20 Sampling frequency: 1000 Hz what should be ...
0
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1answer
39 views

Speaker Recognition

How can you filter out a person's voice from a group of people talking? We have a sample of each person's voice from the group, and the sample of the entire group talking at once. Both samples are ...
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1answer
82 views

Filter design with 8-tap latency on recursion

I'm curious about the feasibility of designing a noise shaping filter (but might generalise to any recursive filter) with the constraint that the most recent output samples aren't available for ...
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1answer
67 views

Given input and output time series, obtaining filter design

Say I have 5 minutes of input and output audio. One method I know is to do FFT on windows from input and output. Then divide FFT output into bins and find average energy in various bands for all the ...
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1answer
102 views

Continuous-time mathematical formula for deconvolution filters

I have an elementary function $p:\mathbb{R}^2\rightarrow\mathbb{R}$ which (locally) represents an image. It's a polynomial, and its the result of the following 2D convolution: $$p=f\star G\star ...
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2answers
109 views

Design Filter Banks for every 100 Hz

I need to study a wave file for every 100 Hz. However, I dont know how can I design filer for each 100 Hz. Let's say 0-100 Hz, 100-200 Hz, ... , ((fs/2)-100) _ (fs/2) Hz. Is it any MATLAB code for ...
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25 views

question about the frequency of Gabor filters

The Fourier transform of 2D Gabor filters is shown below: F is the radial frequency of the Gabor functions. Another parameter Fm is denoted as highest central frequency. Fr is the constant ...
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1answer
80 views

Bireciprocal lattice wave digital filter

I have posted this question "Electrical Engineering", but this seems a more appropiate place. I am trying to model a bireciprocal Cauer filter in LTspice but I don't get the expected results. More ...
0
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2answers
161 views

Bandpass Filter window length: filter “in phase” with sinusoidal input

Say you have a bandpass filter and you place it on 40 discretely sampled points from a sinusoid with a period of 40. Why is the bandpass filter (green curve on the image) only "in phase" with the ...
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1answer
207 views

fir filter - mean delay and snr from difference equation?

I'm trying to get my dsp sea legs a bit, and am trying to complete a problem that asks for the mean delay and expected SNR boost for a given difference equation: y[n] = (x[n] + ... + x[n-N+1])/N I'd ...
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3answers
159 views

Digital filters with more zeros than poles

I am having trouble wrapping my head around digital filters with different orders of numerator and denominator. Let me know if any of these points is wrong: All (digital or analog) transfer ...